If you're using SIP I think what you want is canreinvite=yes which means the two remote user clients can talk directly to each other. Asterisk disappears from the loop which means no accounting. I think NAT causes problems in this scenario also.

More details on the wiki

Regards

Cameron
----- Original Message ----- From: "cmould" <[EMAIL PROTECTED]>
To: <asterisk-users@lists.digium.com>
Sent: Saturday, April 09, 2005 7:48 PM
Subject: [Asterisk-Users] sip phone extensions at a remote site



I am in the proscess of integrating a clients remote and head office
phone systems. Currenty each office has their own PBX and trunk lines. I
am recommending that they put in an Asterisk server at the Head office
with a WAN link to the remote office and switch to IP phones.  Trunk
lines at the remote site would  be returned to the TELCO. External calls
over the PSTN from the remote office would be routed over the WAN to the
head office and through Asterisk to the PSTN trunk lines. All phones
would then become extensions (both remote and head office locations). I
want Person A in the remote office to dial an extension number and get
Person B in the head office. What I am unsure about is if person A and
Person B are both at the remote site and Asterisk PBX is at the head
office, can A and B talk directly to each pther without traversing the
WAN link? Has anyone done this before? What is the quality of the call
if they have? Any information is useful.




--------------------------------------------------------------------------------


_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

_______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Reply via email to